1. Field of the Invention
The present invention relates to a data recording technique for the so-called IP or Internet telephony. More particularly, the invention relates to a data recording system for IP telephony that realizes approximately the same voice or speech communication as the conventional, ordinary telephony by the use of the Internet Protocol (IP) and the IP-based computer network.
With IP telephony, voice or audio data (i.e., audio data) to be transmitted is divided into IP packets (i.e., audio IP packets) and then, these packets are successively sent from a telephone terminal to another distant from it by way of an IP-based computer network. When audio data is transmitted using IP telephony, audio IP packets corresponding to the audio data need to be processed in real time. On the other hand, if audio data is transmitted as it is, it is impossible to ensure a required frequency bandwidth for data transmission. This is because the IP based network is of the so-called xe2x80x9cbest effortxe2x80x9d type. As a result, conventionally, audio data is usually transmitted using the User Datagram Protocol (UDP) and the Realtime Transport Protocol (RTP) for the transport layer of the well-known OSI (Open Systems Interconnection) reference model.
With IP telephony, voice or audio data (i.e., audio data) to be transmitted is divided into IP packets (i.e., audio IP packets) and then, these packets are successively sent from a telephone terminal to another distant from it by way of an IP-based computer network. When audio data is transmitted using IP telephony, audio IP packets corresponding to the audio data need to be processed in real time. On the other hand, if audio data is transmitted as it is, it is impossible to ensure a required frequency bandwidth for data transmission. This is because the Ip-based network is of the so-called xe2x80x9cbest effortxe2x80x9d type. As a result, conventionally, audio data is usually transmitted using the User Datagram Protocol (UDP) and the Realtime Transport Protocol (RTP) for the transport layer of the well-known OSI (Open Systems Interconnection) reference model.
Since IP telephony uses the IP protocol for data transmission, part of the IP packets tend to be lost during transmission and the packets thus lost are automatically resent from the telephone terminal from which the lost IP packets are originated. Generally, the loss rate of the IP packets during transmission varies dependent on the current amount of the traffic on an IP-based network. Thus, there is a problem the voice or speech communication quality is likely to deteriorate.
Moreover, the amount of the traffic on an IP network fluctuates at all times and abrupt increase of the traffic amount is unable to be anticipated. If the fluctuation of the traffic amount can be controlled to an extent by a process such as giving the order of priority to the audio IP packets, the above-described problem about the communication quality deterioration may be suppressed. In this case, however, not only the functions of routers connected to the network but also the entire operation of the network itself need to be additionally controlled. Thus, it is not realistic.
Additionally, even if only the operation of the IP telephone terminal is controlled to give the order of priority to the audio IP packets, it is difficult to make sure that the audio data packets are transmitted through the network as intended.
Furthermore, if the loss rate of the audio packets increases when the amount of traffic is large, the packet retransmission process and/or the congestion control process is/are not performed, where only the real-time process using the RTP protocol is carried out. Therefore, in this case, the speech quality deterioration becomes more conspicuous.
Considering the above-described characteristics of IP telephony, conventionally, a function to complement the lost IP packets during transmission is incorporated into the IP telephone terminal and/or the IP telephone subscriber circuit of an exchange. This lost-packet complementing function is implemented by anticipating the audio data contained in a lost IP packet based on its precedent and subsequent audio data. Therefore, the audio data complemented by this function does not accord perfectly with that contained in the lost packet. Although various researches on the lost-packet complementing function have been conducted, it is unable to realize complete reproduction of the original voice or speech as long as this function is used.
The complement of the lost packet is more difficult if the loss rate of the packets increases furthermore as the traffic increases. This means that in this case, the complemented data is noticeably different from the original one. Thus, the reproduced speech tends to include some sensible distortion.
As explained above, when audio data is transmitted over the IP-based network, the quality of the reproduced speech is affected by the fluctuation of uncontrollable traffic. Thus, to keep the quality degradation and the transmission delay of audio data over the IP network at the same level as the conventional, ordinary telephony by way of the telephone lines, a control method using the TOS (Type of Service) field included in the header of an IP packet may be adopted, for example. However, in this method, high performance routers capable of interpreting the content of the TOS field are required over the whole IP-based network. As a result, this method is difficult to be adopted practically.
Moreover, a voice or sound recording method on a magnetic tape or the like (which has been incorporated in the conventional telephone-answering machines) may be adopted to realize approximately the same quality degradation and approximately the same transmission delay over the IP-based network as the conventional, ordinary telephony. Ordinary telephone-answering machines record directly the voice or speech generated by the handset of a telephone on a magnetic tape. If this direct recording method in the ordinary answering machine is applied to the IP telephone terminal, there is a possibility that the quality of reproduced voice or speech tends to degrade due to the loss of IP packets described previously. As a result, even if the voice or speech generated by the IP telephone terminal is directly recorded on a magnetic tape, the above-identified disadvantage of IP telephony (i.e., the complete reproduction of original voice or speech is impossible) is unable to be solved.
Accordingly, an object of the present invention is to provide a data recording system for IP telephony that complements perfectly the lost audio data due to loss of the IP packets during transmission.
Another object of the present invention is to provide a data recording system for IP telephony that eliminates the necessity of the lost-packet complementing function for the lost audio IP packets.
The above objects together with others not specifically mentioned will become clear to those skilled in the art from the following description.
A data recording system for IP (Internet Protocol) telephony according to the invention comprises;
(a) an IP-based network;
(b) a first telephone terminal connected to the network;
the first telephone terminal being capable of transmission and reception of audio data in the form of IP packets, making communication using an IP;
(c) a second telephone terminal connectable directly to the network or indirectly thereto by way of an exchange;
the second telephone terminal being capable of speech communication; and
(d) a recording device connected to the network;
the recording device being capable of recording audio data transmitted between the first telephone terminal to the second telephone terminal;
wherein when communication is performed between the first telephone terminal and the second telephone terminal, speech IP packets corresponding to audio data are formed and then, the speech IP packets thus formed are transmitted between the first telephone terminal and the second telephone terminal by way of the network in approximately real time; and
wherein recording IP packets corresponding to the audio data are formed and then, the recording IP packets are transmitted to the recording device by way of the network in a way that does not cause any IP packet loss during transmission and that is not performed in real time, thereby recording the audio data by the recording device.
With the data recording system according to the invention, the recording device connected to the IP-based network is provided, where the recording device is capable of recording the audio data transmitted between the first and second telephone terminals. When speech communication is performed between the first and second telephone terminals, the speech IP packets corresponding to the audio data are formed and then, they are transmitted between the first and second telephone terminals by way of the network in approximately real time. On the other hand, the recording IP packets corresponding to the audio data are formed and then, the recording IP packets are transmitted to the recording device by way of the network in the way that does not cause any IP packet loss during transmission and that is not performed in real time. The audio data thus transmitted is recorded by the recording device.
Thus, communication is performed with the speech IP packets transmitted over the network in real time. At the same time as this, the audio data transmitted by the recording IP packets is recorded by the recording device, where the recording IP packets are not lost during transmission and the transmission is not carried out in real time.
Accordingly, if the audio data stored in the recording device is read out for reproduction after the communication is finished, all the audio data transmitted by the speech IP packets without IP packet loss over the network can be reproduced. In other words, the necessity of the lost-packet complementing function for the lost audio IP packets is eliminated, which realizes defect-free speech or voice communication between the first and second terminals.
In a preferred embodiment of the system according to the invention, the recording IP packets are transmitted to the recording device using the Transmission Control Protocol (TCP) for the transport layer of the OSI reference model. In this embodiment, the recording IP packets are transmitted to the recording device using the TCP that conducts error-recovering processes during transmission such as an automatic packet-resending process. Thus, defect-free speech/voice communication can be easily realized without the use of the lost-packet complementing function.
In another preferred embodiment of the system according to the invention, the speech IP packets are transmitted over the network using the UDP and the RTP. In this embodiment, the speech IP packets are transmitted over the network using the UDP and the RTP and therefore, they are transmitted over the network in approximately real time like the conventional IP telephony. If some of the speech IP packets are lost during transmission, the lost IP packets are complemented by anticipating the audio data contained in the lost IP packets based on their precedent and subsequent audio data, thereby enabling the substantially real-time speech communication.
In still another preferred embodiment of the system according to the invention, the recording device is designed to be recordable only when the first telephone terminal is in an off-hook state. In this embodiment, the recording device does not record the recording IP packets when the first telephone terminal is in an on-hook state and therefore, there is an additional advantage that the running cost of the system decreases surely.
In a further preferred embodiment of the system according to the invention, the recording device starts its recording operation based on a recording start order emitted from the first telephone terminal. In this embodiment, the recording device starts its recording operation according to the intention of the user of the first telephone terminal and therefore, a necessary part of the communication is selectively recorded. Thus, there is an additional advantage that the limited volume of a recording medium of the recording device is efficiently used.
In a still further preferred embodiment of the system according to the invention, the recording device is designed to finish its recording operation by an on-hook operation of the first telephone terminal. In this embodiment, there is an additional advantage that the recording operation of the device is surely prevented in the on-hook state of the first telephone terminal after the communication is finished.
In a still further preferred embodiment of the system according to the invention, the recording device finishes its recording operation based on a recording stop order emitted from the first telephone terminal. In this embodiment, the recording device finishes its recording operation according to the intention of the user of the first telephone terminal and therefore, a necessary part of the communication is selectively recorded. Thus, there is an additional advantage that the limited volume of a recording medium of the recording device is efficiently used.
In a still further preferred embodiment of the system according to the invention, the recording device is designed to record in such a way that the speech IP packets from the first telephone terminal and those from the second telephone terminal or the exchange are independent from each other. In this embodiment, there is an additional advantage that the audio data from the first terminal and that from the second terminal or the exchange can be reproduced separately. Thus, even if the audio data from the first terminal and that from the second terminal or the exchange are overlapped and difficult to be heard, they can be clearly heard by reproducing them separately.
In this embodiment, preferably, the recording device is designed to record a real time value of a first one of the recording IP packers from the first telephone terminal and a real time value of a first one of the recording IP packets from the second telephone terminal or the exchange. In this case, there is an additional advantage that the time sequence of the recorded audio data from the first terminal and the time sequence of the recorded audio data from the second terminal or the exchange can be identified separately.
In this case, preferably, the recording device is designed to reproduce the recorded audio data from the first telephone terminal and the recorded audio data from the second telephone terminal or the exchange according to their real time sequence, respectively. In this case, there is an additional advantage that the voice or speech communication between the first and second telephone terminals can be entirely reproduced with the recording device.
In a still further preferred embodiment of the system according to the invention, the recording device is formed by an information-processing device (e.g., a personal computer or server computer) connected to the network. In this embodiment, there is an additional advantage that the audio data is easily recorded and easily reproduced, because a storage device incorporated into a computer serving as the recording device can be used for recording and reproduction of the audio data.
In a still further preferred embodiment of the system according to the invention, the first telephone terminal and the recording device are united together. In this embodiment, there is an additional advantage that the recording and reproducing operations of the audio data corresponding to the recording IP packets are easily performed if the user operates the first telephone terminal. There is another additional advantage that if the first telephone terminal is formed by software running on an information-processing device, the recording device can be formed by a storage device of the information-processing device.
In the system of the invention, the second telephone terminal may be connected indirectly to the network by way of the exchange. In this case, the second telephone terminal is capable of voice or speech communication with the first telephone terminal by way of the exchange and the network. On the other hand, the second telephone terminal may be connected directly to the network. In this case, the second telephone terminal is capable of transmission and reception of the audio data in the form of IP packets, making voice or speech communication using the IP.